This invention relates to an audio decoding device for expanding audio data transmitted or recorded on a recording medium in a compressed form upon reproduction of the audio signal, or to an audio coding/decoding system for transmitting audio data, or recording same on a recording medium, in a compressed form, and for reproducing the audio data in the expanded form.
There are various known methods for coding audio signals. One of them converts audio signals by using time/frequency conversion that converts a time region signal into a frequency region signal. Used for time/frequency conversion is, for example, a sub-band filter or MDCT (Modified Discrete Cosine Transform).
General information on sub-band filter coding and MDCT coding are given by, for example, Furui & Sondhi in "Advances in Speech Signal Processing" pp. 109-140, published by Marcel Dekkar (New York) in 1991. Known as a sub-band filter coding system is ISO/IEC 11172-3 which is an international standard called MPEG Audio System, and as a MDCT coding system is AC-3 coding.
FIG. 11 shows a conventional audio coding device.
In FIG. 11, a digital audio signal introduced into an input terminal 31 is converted from a time region signal into a frequency region signal in predetermined intervals of time (the interval of time is hereinbelow referred to as conversion block length) by a time/frequency converting circuit 32, and divided into a plurality of frequency bands to increase the coding efficiency.
The converted frequency-region audio signal is supplied to a quantizing circuit 33 for floating and quantization for individual frequency bands therein. Floating herein pertains to a processing for increasing the value of the effective portion of data by multiplying each data in each divisional band by a common value for up-carrying or down-carrying in order to increase the accuracy of subsequent quantization. Floating is not done when quantization accuracy is immaterial. Apractical example of floating is configured to find one having a largest absolute value among data in each band and to use a floating coefficient to maximize the value within the limit not saturating, i.e., not exceeding "1". FIG. 12 shows examples of floating coefficients used in the ISO/ICE 11172-3 system.
The coding device of FIG. 11 executes floating by using an appropriate value among the floating coefficients of FIG. 12. For example, if the maximum absolute value of data in a frequency band is 0.75, then the device selects 0.79370052598410 as a floating coefficient, which is one of floating coefficients of FIG. 12 and whose reciprocal multiplied by 0.75 is maximum within the limit not exceeding "1", and performs floating by multiplying each data in the bands by the reciprocal of the floating coefficient.
The floating coefficient used in the coding device is actually represented and transmitted by a corresponding index value ("4" in the above example). That is, the index value "4" as a floating coefficient selected for floating by the quantizing circuit 33 is transmitted to a multiplexing circuit 34. For decoding, the same floating coefficient is used among those of FIG. 12.
The digital audio signal introduced to the input terminal 31 is supplied also to an adaptive bit assigning circuit 35. The circuit 35 calculates characteristics of an input signal and determines the number of bits to be assigned for each frequency band in accordance with the signal characteristics. For example, the assigned bit number for each frequency band is determined to vary the quantization accuracy adaptively to inaudibilities by the human acoustic sense.
Known as characteristics of the human acoustic sense are minimum audible characteristics which indicate that low frequency sounds are difficult for persons to hear when the volume level is low because the human acoustic sense is lower in low frequency bands, for example, and masking characteristics which indicate the acoustic sense decreases for frequencies near the peak of a certain frequency spectrum.
The human acoustic sense is used for bit assignment to reduce the entire amount of information by modeling audibilities and inaudibilities for individual frequency bands and by assigning less bits to relatively inaudible frequency components.
The assigned bit number determined by the adaptive bit assigning circuit 35 is output as bit length information to the quantizing circuit 33. The quantizing circuit 33 executes quantization of data after floating, using adaptive bit lengths for individual frequency bands. The quantized audio data from the quantizing circuit 33, floating coefficient and bit length information are multiplexed in the multiplexing circuit 34, and output as coded data from an output terminal 37.
FIG. 13 shows a conventional audio decoding device for expanding the compressed audio data from the audio coding device shown in FIG. 11. FIG. 14 is a diagram showing an audio data decoding circuit 51 contained in FIG. 13 in greater detail.
In FIG. 13, the coded audio data supplied to an input terminal 1 is introduced to the audio data decoding circuit 51. As shown in FIG. 14, the coded audio data enters into a demultiplexing circuit 11 at the input stage of the audio data decoding circuit 51. The demultiplexing circuit 11 divides the multiplexed signals for respective frequency bands into audio data, floating coefficient and bit length information for each band.
The divided audio data is supplied to a dequantizing circuit 12 for dequantization and inverse-floating for each frequency band. Quantization is done using the bit length information for each frequency component divided by the demultiplexing circuit 11. Inverse-floating is done for dequantized data in each frequency band by multiplying the dequantized audio data by the floating coefficient divided by the demultiplexing circuit 11, which is one of index values shown in FIG. 12.
The audio data after dequantization and inverse-floating in the dequantizing circuit 12 is converted from the frequency-region signal into the time-region signal by a frequency/time converting circuit 14. The decoded digital audio signal in form of the time-region signal is output from an output terminal 15 and supplied to a subsequent digital-to-analog converter circuit 3.
The digital audio signal recomposed in the audio data decoding circuit 51 is converted into an analog signal by a digital-to-analog converter circuit 3, then adjusted in volume level by a volume control circuit 4, passed through an output adjusting circuit 52, and output from an output terminal 5. Volume adjustment is done by a user of the audio decoding device as desired through a volume knob or other element, not shown.
As explained above, the human acoustic sense has the nature that low frequency components are difficult to hear when the volume is low. Therefore, when audio signals are reproduced in a low volume, they sound as lacking low frequency components, and give a bad quality of sound to human ears. To remove such phenomenon, the output adjusting circuit 52 makes adjustment to enhance low frequency components depending on information on the selected output volume.
U.S. Pat. No. 4,739,514 discloses a sort of the output adjusting circuit 52. This patent uses a band pass filter for dynamically adjusting low frequency components by analog processing to its time-region signal. This circuit, however, needs a number of operational amplifiers and other analog circuit elements, and inevitably becomes a large-scaled and complex circuit.
The human acoustic sense involves the nature that also high frequency components, in addition to low frequency components, are difficult to hear during reproduction in a low volume level. The above-indicated patent, however, makes adjustment of low frequency components alone. Without adjustment of high frequency components, the quality of sound, as a whole, remains bad even after adjustment of low frequency components.
Although conventional techniques use human acoustic characteristics for bit assignment, it enhances low frequency components in the output adjusting circuit 52 upon reproduction irrespectively of the nature of the original signal components, and causes the reproduced signal to have a property different from the acoustic sense model calculated during coding. As a result, enhanced lowband quantized noise is heard, and hence damages the quality of sound to human ears.